- Calculate size for both formats (+ move the return to transcode_encode_query)
- Let transcode_needed() decide what format to output
- Determine content-type from transcoding type
- Add transcode-dependent ability to override file metadata in rsp/daap
- Send file size matching format
ffmpeg 6 requires fixed size frames for the encoder, so this commit refactors
the mp3 streaming to do support that. Also uses outputs_quality_subscribe which
potentially has better performance (e.g. a 48000 sample rate source will only
be resampled once if both airplay and mp3 streaming want 44100), and makes it
easier to create fixed size frames.
Resolves#1601
Roku SoundBridge devices can be treated as dumb speakers; they offer a
texted based protocol, RCP, that allows the server to request the Roku
device to connect to the server's stream.mp3 and provides ability to
control volume
Roku devices appear as speakers automatically via mDNS announcements on
the server _roku-rcp._tcp
Includes
- Implementation in src/outputs/airplays2, type OUTPUT_TYPE_AIRPLAY
- Homekit pairing, both normal (with PIN) and transient
- New session startup sequence, incl GET /info, SETPEERS and 2 x SETUP
- No more OPTIONS and ANNOUNCE
- Use POST /feedback for keepalive instead of SET_PARAMETERS
- Sequence dispatching instead of callback chains
- Continue despite "Bad request" to SET_PARAMETER (volume)
- Opening of event connection to receiver (reverse rtsp connection)
Still to be done
- Password authentication
- Handling of events
outputs_master_volume, output_device_list and output_buffer were defined in
outputs.h, but should only have been declared with extern.
Solve it by making output_buffer static/non-global (wasn't used anyway), and
the two others are now returned by functions.
* Make sure that vol_adjust() bases adjustments on the same speakers that the
user sees as active
* If a speaker fails during activation we unselect it, so that we don't keep
trying to start it, so we find a new master volume if required, and so it is
possible for the user to reselect it.
outputs_authorize() has two issues, one that the caller can't specify device
(problem if there are two devices waiting for verification), the other that it
didn't offer a standard callback, so difficult to catch failure/success.
Moves speaker selection, volume handling and startup to outputs.c, plus adds
the ability to "resurrect" a speaker that disconnects.
The purpose of moving the code is to concentrate device handling in one place.
Also changes how we deal with speaker selection. The player will now generally
not alter a user selection, even if the device fails. The purpose of this is to
maintain selection both if the device briefly fails, and if the user switches
off the device (we stop playback) and later turns it on + starts new playback.
If the number of quality subscriptions reaches max then this bug will be
triggered, because we will incorrectly use the last element of the
output_buffer for a subscription, thus losing the zero terminator.
This change is preparation to use ffmpeg's resampling capabilities to keep local
audio in sync (by up/downsampling slightly). This requires that sample rates are
not fixed for a transcode profile.
Added benefit of this is that we don't need quite as many xcode profiles.
In the output implementations playback_stop() was somewhat redundant,
since device_stop() does the same.
The timer should make sure that we always close outputs (previously
they were in some cases kept open).
The commit also includes some renaming.
* Drop output_sessions, was just a pointer to the actual session anyway
* Drop the old write, flush and stop prototypes
* Some minor changes/renaming
Purpose of this is also to fix a race condition in player.c where it
could try to start two sessions on the same speaker. This could happen
because outputs_device_start() in line 2093 is conditional on device->session
which however is false while a device is starting up.
outputs_playback_start() had the problem that was not consistently invoked: If
for instance local audio playback was running and a Airplay device was then
activated, the raop's playback_start would never be invoked (and vice versa,
of course).
Instead, the player now writes the presentation timestamp every time to the
output, so it doesn't need to keep track of it from the start.
Output module can now take input data in multiple quality levels, and
can resample to those output modules that would require a certain quality
level, like raop.c would
Implements Active-Remote, which is sent to the speaker, so it can use this
to tell us who it is when it makes dacp request with a device-volume update.
- also change how speakers are saved/retrieved from the db
- add generic authorization methods in outputs.c and player.c
- let filescanner read *.verification files (containing PIN)
- configure options to enable and disable, since libsodium is required
- Added custom checks for libraries and pkgconfig modules that test library
presence with additional checks for use of headers and functions with
given options. Also support correct additional feature library checks
using provided flags.
- Added custom enable/disable feature macros to simplify their use.
- Use custom CFLAGS and LIBS variables for Makefiles to simplify maintenance.
- Update many feature checks from platform to function.
- Streamline many function checks.
- Correctly check gnutls, gcrypt and gpg-error libraries.
- Fix chromecast and spotify config and compile on FreeBSD
- Added inotify, signalfd and kqueue, and byte swap checks.
- Many clarifications of error messages.
- Correct json-c checks to properly use supplied CFLAGS.
- Correct many quoting inconsistencies
- Use __DATE__ in place of BUILDDATE
- Use full path for gperf and antlr3
- Remove unnecessary CFLAGS
- Added tests for pthread_setname_np parameters
- Added tests for clock_gettime and timer_settime
- Added tests for time.h
- Test if pthread, dl and rt libs are required/available.
- Updated checks for libunistring
